What is FreePBX and how to connect it as a SIP client

FreePBX: what it is and what it is used for

FreePBX is a popular web interface for setting up and managing Asterisk-based phone systems. 

The main features of FreePBX:

• Management of incoming and outgoing calls.
• Support for different telephone lines (SIP, PJSIP, Analog, ISDN).
• Ability to create and manage voice menus (IVR).
• Easy integration with other services and protocols such as SIP.
• User-friendly web interface for configuration and monitoring.

The following are the basic features of Free PBX that are available to users and make it a popular service for setting up VoIP and traditional phone systems:

1. Extensions.
   

   FreePBX allows you to configure and manage telephone extensions (SIP, PJSIP, IAX). You can create new extensions, assign passwords to them, and configure parameters for each device or user. This can be either a hardware phone or a software SIP client.

2. Voicemail.
   

   FreePBX provides functionality to configure voicemail boxes for each extension. Subscribers can receive voicemails, listen to them via phone, email, or web interface, and set up notifications of new messages.

3. Interactive Voice Response (IVR).
   

   IVR (Interactive Voice Response) allows you to set up a voice menu to route calls. Users can select different options to get to the right department or group (e.g., “For sales, press 1; for technical support, press 2”). This improves customer service by automating calls.

4. Call Routing.
   

   FreePBX allows you to configure routes for incoming and outgoing calls, directing them in different scenarios. For example, calls can be routed to queues, voice menus, or depending on the time of day. You can set up different routes for weekends and weekdays.

5. Queues.
   

   FreePBX supports the creation of queues to handle incoming calls. Queues allow you to efficiently distribute calls between groups of agents, ensuring proper routing based on agent availability.

6. Call Recording.
   

   FreePBX provides a feature to record all calls, both incoming and outgoing. The recordings can be saved for later analysis, employee training, or for legal compliance. You can set up automatic or manual call recording.

7. Conference.
   

   FreePBX allows you to organize conference calls, allowing multiple users to communicate in the same session. This can be useful for meetings, conferencing, or supporting teamwork. Administrators can configure conference settings, including PIN codes for accessing and managing participants.

8. Auto Attendant.
   

   Auto Attendant is a feature that allows you to automatically answer calls with a pre-recorded message. It can be used for initial call routing or to provide information to the caller without operator intervention.

9. Trunks.
   

   FreePBX supports integration with various external telephone lines such as SIP, IAX, PRI, analog, and others. This allows you to connect phone lines from external providers and use them for outgoing and incoming calls.

10. Reports & Monitoring.
    

    FreePBX provides tools to monitor and report on the performance of your telephone system. This includes call statistics, reports on call duration and cost, and system logs to track errors and performance. This data helps to improve management and monitor system performance.

What is a SIP client?

SIP (Session Initiation Protocol) is a protocol that is used to initiate, maintain, and terminate multimedia communication sessions such as voice calls or video conferencing over the Internet. A SIP client is software or a device that supports this protocol and can be used to make or receive calls over an IP network.

If you want to use the FreePBX as a SIP client, this means that the FreePBX will connect to an external SIP server or VoIP provider to make calls or to integrate with other telephone systems.

How to perform the default configuration of FreePBX as a SIP client

Before starting the configuration you need to create a SIP account in Maxnet's personal account. 

    1. Specify the trunk name and go to the “General” tab.

    2. Click the “pjsip setting” tab.

    3. In the “PEER Details” and “USER Details” sections, provide the following details as in the example:

• username and auth username is the login from the number;
• secret is its password;
• Authentication — select outgoing;
• Registration — Send;
• sip server — vproxy2.maxnet.ua;
• Port — 5060;
• Context — depends on the settings of your FreePBX, by default, you can leave it like this;
• Transport — UDP.

    4. In the “Connectivity” — “Inbound Routes” section, create a route for incoming calls.

    5. In the “Description” section specify “_57720353” (as in the example).

    6. In the “Set Destination” section you can specify where the incoming call will be addressed (FreePBX extension number, call group, IVR, etc.).

    7. Go to “Connectivity” — “Outbound Routes” and create a route for outbound calls.

To do this, select a previously created trunk, as in the example: 

• Route Name: _57720353;
• Trunk Sequence for Matched Routes: 507209353.

    8. In the “Dial Patterns” section, in the “match pattern” field, set a point (if you do not set a point, you will not be able to make outgoing calls) and create a route.

This completes the standard configuration of FreePBX.

Our team of specialists is ready to connect a Virtual PBX based on Asterisk FreePBX to solve the problems of your business. You can conveniently order the service through the Maxnet control panel.